git.sesse.net Git - mlt/blob - src/modules/resample/filter_resample.c

   1 /*
   2  * filter_resample.c -- adjust audio sample frequency
   3  * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
   4  * Author: Dan Dennedy <dan@dennedy.org>
   5  *
   6  * This program is free software; you can redistribute it and/or modify
   7  * it under the terms of the GNU General Public License as published by
   8  * the Free Software Foundation; either version 2 of the License, or
   9  * (at your option) any later version.
  10  *
  11  * This program is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  14  * GNU General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU General Public License
  17  * along with this program; if not, write to the Free Software Foundation,
  18  * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
  19  */
  20
  21 #include <framework/mlt_filter.h>
  22 #include <framework/mlt_frame.h>
  23
  24 #include <stdio.h>
  25 #include <stdlib.h>
  26 #include <samplerate.h>
  27 #define __USE_ISOC99 1
  28 #include <math.h>
  29
  30 #define BUFFER_LEN 20480
  31 #define RESAMPLE_TYPE SRC_SINC_FASTEST
  32
  33 /** Get the audio.
  34 */
  35
  36 static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
  37 {
  38         // Get the properties of the frame
  39         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
  40
  41         // Get the filter service
  42         mlt_filter filter = mlt_frame_pop_audio( frame );
  43
  44         // Get the filter properties
  45         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
  46
  47         // Get the resample information
  48         int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
  49         SRC_STATE *state = mlt_properties_get_data( filter_properties, "state", NULL );
  50         float *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
  51         float *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
  52         int channels_avail = *channels;
  53         SRC_DATA data;
  54         int i;
  55
  56         // If no resample frequency is specified, default to requested value
  57         if ( output_rate == 0 )
  58                 output_rate = *frequency;
  59
  60         // Get the producer's audio
  61         mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
  62
  63         // Duplicate channels as necessary
  64         if ( channels_avail < *channels )
  65         {
  66                 int size = *channels * *samples * sizeof( int16_t );
  67                 int16_t *new_buffer = mlt_pool_alloc( size );
  68                 int j, k = 0;
  69
  70                 // Duplicate the existing channels
  71                 for ( i = 0; i < *samples; i++ )
  72                 {
  73                         for ( j = 0; j < *channels; j++ )
  74                         {
  75                                 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
  76                                 k = ( k + 1 ) % channels_avail;
  77                         }
  78                 }
  79
  80                 // Update the audio buffer now - destroys the old
  81                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
  82
  83                 *buffer = new_buffer;
  84         }
  85         else if ( channels_avail == 6 && *channels == 2 )
  86         {
  87                 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
  88                 int size = *channels * *samples * sizeof( int16_t );
  89                 int16_t *new_buffer = mlt_pool_alloc( size );
  90
  91                 // Drop all but the first *channels
  92                 for ( i = 0; i < *samples; i++ )
  93                 {
  94                         new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
  95                         new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
  96                 }
  97
  98                 // Update the audio buffer now - destroys the old
  99                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
 100
 101                 *buffer = new_buffer;
 102         }
 103
 104         // Return now if no work to do
 105         if ( output_rate != *frequency )
 106         {
 107                 float *p = input_buffer;
 108                 float *end = p + *samples * *channels;
 109                 int16_t *q = *buffer;
 110
 111                 // Convert to floating point
 112                 while( p != end )
 113                         *p ++ = ( float )( *q ++ ) / 32768.0;
 114
 115                 // Resample
 116                 data.data_in = input_buffer;
 117                 data.data_out = output_buffer;
 118                 data.src_ratio = ( float ) output_rate / ( float ) *frequency;
 119                 data.input_frames = *samples;
 120                 data.output_frames = BUFFER_LEN / *channels;
 121                 data.end_of_input = 0;
 122                 i = src_process( state, &data );
 123                 if ( i == 0 )
 124                 {
 125                         if ( data.output_frames_gen > *samples )
 126                         {
 127                                 *buffer = mlt_pool_realloc( *buffer, data.output_frames_gen * *channels * sizeof( int16_t ) );
 128                                 mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, mlt_pool_release, NULL );
 129                         }
 130
 131                         *samples = data.output_frames_gen;
 132                         *frequency = output_rate;
 133
 134                         p = output_buffer;
 135                         q = *buffer;
 136                         end = p + *samples * *channels;
 137
 138                         // Convert from floating back to signed 16bit
 139                         while( p != end )
 140                         {
 141                                 if ( *p > 1.0 )
 142                                         *p = 1.0;
 143                                 if ( *p < -1.0 )
 144                                         *p = -1.0;
 145                                 if ( *p > 0 )
 146                                         *q ++ = 32767 * *p ++;
 147                                 else
 148                                         *q ++ = 32768 * *p ++;
 149                         }
 150                 }
 151                 else
 152                         fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
 153         }
 154
 155         return 0;
 156 }
 157
 158 /** Filter processing.
 159 */
 160
 161 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 162 {
 163         if ( mlt_frame_is_test_audio( frame ) == 0 )
 164         {
 165                 mlt_frame_push_audio( frame, this );
 166                 mlt_frame_push_audio( frame, resample_get_audio );
 167         }
 168
 169         return frame;
 170 }
 171
 172 /** Constructor for the filter.
 173 */
 174
 175 mlt_filter filter_resample_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
 176 {
 177         mlt_filter this = mlt_filter_new( );
 178         if ( this != NULL )
 179         {
 180                 int error;
 181                 SRC_STATE *state = src_new( RESAMPLE_TYPE, 2 /* channels */, &error );
 182                 if ( error == 0 )
 183                 {
 184                         void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
 185                         void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
 186                         this->process = filter_process;
 187                         if ( arg != NULL )
 188                                 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "frequency", atoi( arg ) );
 189                         mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
 190                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
 191                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 192                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 193                 }
 194                 else
 195                 {
 196                         fprintf( stderr, "filter_resample_init: %s\n", src_strerror( error ) );
 197                 }
 198         }
 199         return this;
 200 }